As telecommunication networks have evolved, different types of signals are now communicated over telephone lines in addition to voice signals such as DTMF tones, facsimile signals, etc. Although DTMF tones are traditionally generated to initiate a call, they may also be generated during a call, e.g., when a telephone subscriber is entering a bank account number, credit card number, etc. One problem resulting from these different types of machine-generated signals communicated over the telephone network is the difficulty in setting an amplitude/volume level in a telephone receiver that is appropriate for all types of signals. In particular, a signal level optimal for hearing incoming voice signals is typically too high of a signal level for machine-generated tones and other non-voice signals. This particular problem is especially troublesome for telephone operators who employ a headset rather than a handset to place and receive telephone calls. If such an operator receives a DTMF tone or other machine-generated tone at the volume level set for listening to voice, it is often not possible for the operator to remove the headset in time to avoid "shocking,", or even worse, damaging the operator's ears.
Machine-generated noise like DTMF and fax tones occur unexpectedly and for short time durations making them difficult to anticipate and ameliorate. Because such tones can be generated over a relatively wide frequency band that overlaps the voice bandwidth, it is expensive and therefore not practical to employ fixed filters at particular frequencies to attenuate them. Signal "jamming" of voice communications using periodic interference also presents a problem of changing frequencies. Because such changing periodic interference may be used to "sweep" the voice spectrum, even well-designed filters tailored for a particular and narrow frequency band would be completely ineffective.
Simply lowering the volume level output to a headset is not a particularly good option because the signal level of received speech would likely be too low for a listener to easily discern voice information communicated over the telephone line. Although it might be possible to implement a squelch or noise blanking circuit to squelch/blank all received signals when a tone is detected, such an approach has a major disadvantage. Namely, voice signals or other desired information received at the same time as the tone or other interference would be squelched or blanked out as well.
It is an object of the present invention to provide an audio signal processing method and apparatus that solves these problems by reducing periodic interference in received voice signals so the periodic interference does not (1) surprise the human operator listening to the audio signal using a communications device like a telephone headset or handset or (2) adversely affect the hearing of the human operator.
It is a further object of the present invention to provide such a method and apparatus that specifically reduces periodic interference like facsimile tones, DTMF tones, and other oscillatory signals that may be received during a telephone call being listened to by a human operator.
It is a further object of the present invention to adaptively track and reduce periodic interference that changes in frequency.
It is another object of the present invention to preserve the voice received along with the periodic interference after that periodic interference is reduced or otherwise eliminated.
These and many other objects are achieved by the inventive method which protects an operator listening to electronically-generated audible signals. Received audio signals including both voice signals and interfering signals in the same frequency band are processed before being sent to a speaker. Periodic interference is modeled, and based on that modeled periodic interference, the actual periodic interference in the received audio signals is reduced or removed. As a result, the periodic interference does not adversely affect the hearing of the operator listening to the received audible signals. Moreover, the operator still readily discerns the voice received along with the periodic interference despite the fact that the interference has been reduced or removed. Thus while the voice suffers little or no attenuation, the periodic interference is selectively and substantially attenuated.
The present invention has particularly advantageous application in a call handling system such as a centralized call handling center where plural telephone operators are connected to a telecommunications switch via a network or other routing mechanism. Each operator is provided with a computer station, such as a personal computer, along with a communications device having a microphone and a speaker. Audio signals are received from the telecommunications switch for delivery to the communications device including both voice and is periodic interference. The volume or decibel level of the periodic interference may exceed a threshold that is uncomfortable or otherwise dangerous to the operator listening to the received signal.
The received signals are processed to make sure that this periodic interference does not exceed that threshold. The audio signal is sampled at a predetermined sampling rate to generate a signal sequence. Based on a previous sequence of samples, a contribution of a current signal sample corresponding to the periodic interference is predicted. That predicted contribution is removed from the current sample to ensure that the decibel level of the periodic interference present in the current signal sample ultimately used in reconstructing the analog signal delivered to the speaker of the communications device is less than the threshold.
Periodic interference is predicted by taking advantage of its periodicity. The interference is predicted directly from the audio signal without the need for an external reference signal (corresponding to the interference) generated from a source other than the audio signal itself. More specifically, the signal sequence is processed by delaying the signal sequence for a predetermined delay. The delayed signal sequence is filtered, and the filter output consisting of an estimated current value of the periodic interference sequence is subtracted from an undelayed current value of the signal sequence to generate a difference signal. The periodic interference has therefore been substantially attenuated or effectively removed or cancelled. In this way, the current value of the periodic interference sequence is predicted using previous values of the received signal sequence. The predicted signal is removed from the actual audio signal to generate an audio signal that for the most part is includes only the desired voice signal.
In such a call handling system, each telephone operator's work station (or communication device) contains an audio processor. The audio processor includes an analog-to-digital converter for converting received audio signals into digitized audio samples. An adaptive filter receives the digitized audio samples and substantially reduces the amplitude of tones or other periodic interference in a current audio sample while at the same time substantially preserving both the content and the signal level of the voice information in the current audio sample. Thereafter, a digital-to-analog convertor converts the signal output from the adaptive filter into an analog audio signal. An amplifier may be provided to amplify the audio signal and then forward the amplified signal to drive the speaker in the operator's communication device.
The audio processor implements a signal processing procedure that predicts the tones or other periodic interference occurring in the current audio sample based on previously processed audio samples and subtracts the predicted tones or other periodic interference from the current sample. The predictor is adaptive and therefore tracks changes in frequency in the tones or other interference.
The present invention further provides a digital filter which provides particularly advantageous application in a call handling system such as that described above but also has wide signal processing application. The digital filter includes a delay stage that receives a digitized sequence of samples corresponding to an audio signal that includes both voice and interference. Series-connected filter taps are connected to the delayed signal sequence. Parallel multipliers each multiply a signal from a corresponding one of the filter taps and a corresponding adaptive filter coefficient. A summer sums outputs from the multipliers to generate a signal that predicts the periodic interference in the current signal sample. The predicted periodic interference is then subtracted from the current sample to obtain a difference signal corresponding substantially or entirely to just the voice content of the input audio signal. That difference signal is also employed by an adaptive controller that recursively modifies the filter coefficients in order to minimize the error between the predicted periodic interference and the actual periodic interference. The optimal parameters of the digital filter for application in a call handling system such as that described above are disclosed.